1. Field of the Invention
This invention relates to audio transducer compensation, and more particularly to a method of compensating linear and non-linear distortion of an audio transducer such as a speaker, microphone or power amp and broadcast antenna.
2. Description of the Related Art
Audio speakers preferably exhibit a uniform and predictable input/output (I/O) response characteristic. Ideally, the analog audio signal coupled to the input of a speaker is what is provided at the ear of the listener. In reality, the audio signal that reaches the listener's ear is the original audio signal plus some distortion caused by the speaker itself (e.g., its construction and the interaction of the components within it) and by the listening environment (e.g., the location of the listener, the acoustic characteristics of the room, etc) in which the audio signal must travel to reach the listener's ear. There are many techniques performed during the manufacture of the speaker to minimize the distortion caused by the speaker itself so as to provide the desired speaker response. In addition, there are techniques for mechanically hand-tuning the speaker to further reduce distortion.
U.S. Pat. No. 6,766,025 to Levy describes a programmable speaker that uses characterization data stored in memory and digital signal processing (DSP) to digitally perform transform functions on input audio signals to compensate for speaker related distortion and listening environment distortion. In a manufacturing environment, a non-intrusive system and method for tuning the speaker is performed by applying a reference signal and a control signal to the input of the programmable speaker. A microphone detects an audible signal corresponding to the input reference signal at the output of the speaker and feeds it back to a tester which analyzes the frequency response of the speaker by comparing the input reference signal to the audible output signal from the speaker. Depending on the results of the comparison, the tester provides to the speaker an updated digital control signal with new characterization data which is then stored in the speaker memory and used to again perform transform functions on the input reference signal. The tuning feedback cycle continues until the input reference signal and the audible output signal from the speaker exhibit the desired frequency response as determined by the tester. In a consumer environment, a microphone is positioned within selected listening environments and the tuning device is again used to update the characterization data to compensate for distortion affects detected by the microphone within the selected listening environment. Levy relies on techniques for providing inverse transforms that are well known in the field of signal processing to compensate for speaker and listening environment distortion.
Distortion includes both linear and non-linear components. Non-linear distortion such as “clipping” is a function of the amplitude of the input audio signal whereas linear distortion is not. Known compensation techniques either address the linear part of the problem and ignore the non-linear component or vice-versa. Although linear distortion may be the dominant component, non-linear distortion creates additional spectral components which are not present in the input signal. As a result, the compensation is not precise and thus not suitable for certain high-end audio applications.
There are many approaches to solve the linear part of the problem. The simplest method is an equalizer that provides a bank of bandpass filters with independent gain control. More elaborate techniques include both phase and amplitude correction. For example, Norcross et al “Adaptive Strategies for Inverse Filtering” Audio Engineering Society Oct. 7-10, 2005 describes a frequency-domain inverse filtering approach that allows for weighting and regularization terms to bias an error at some frequencies. While the method is good in providing desirable frequency characteristics it has no control over the time-domain characteristics of the inverted response, e.g. the frequency-domain calculations can not reduce pre-echoes in the final (corrected and played back through speaker) signal.
Techniques for compensating non-linear distortion are less developed. Klippel et al, ‘Loudspeaker Nonlinearities—Causes, Parameters, Symptoms’ AES Oct. 7-10, 2005 describes the relationship between non-linear distortion measurement and nonlinearities which are the physical causes for signal distortion in speakers and other transducers. Bard et al “Compensation of nonlinearities of horn loudspeakers”, AES Oct. 7-10, 2005 uses an inverse transform based on frequency-domain Volterra kernels to estimate the nonlinearity of the speaker. The inversion is obtained by analytically calculating the inverted Volterra kernels from forward frequency domain kernels. This approach is good for stationary signals (e.g. a set of sinusoids) but significant nonlinearity may occur in transient non-stationary regions of the audio signal.